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Set up Obi212 as FXO for FreePBX, intermittent incoming audio issue

Started by Nmennear, March 28, 2019, 09:53:30 PM

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Nmennear

I'm in the process of rolling out a few FreePBX systems using OBi212s as FXO gateways. So far I've followed the instructions in the sticky and am able to successfully make & receive calls via the OBi212, but I am having persisting issues with incoming call audio dropping for 5-10 seconds intermittently. It seems to only be happening for the incoming call audio (from the PSTN) as the outside caller can hear me throughout the outage.

MY testing so far has all been done locally within the LAN and I've disabled SIP ALG in my router in the event it was causing issues. I've upgraded my OBi's to 3.2.2 firmware and have tried various settings minor settings in the Obi without any success.

I am having the same issue at every location so i suspect there is an important (and hopefully obvious) setting i am missing somwehere. My configuration at each location is identical - FreePBX installed on a physical Gigabyte BRIX, Ubiquiti Edgerouter, OBi212, netgear gigabit switches / WAP.

I've noticed that the packet counters on the OBi continue to show traffic during the dropped audio so it may be something on the FreePBX-side or network related.

Not sure what additional info I can add that would be helpful. Sorry for my ignorance, I've been trying to figure this out for the better part of the week and am unsure where to look at this point.

Thank you very much for any help you can provide!

Nmennear

Just to add to my previous post..

A chan_sip trunk is configured to Asterisk using UDP.

Obi box is configued as bridged and a LAN cable from in place Netgear gigabit switch is connected to the WAN port on the Obi 212.

Settings we have tried:
- Using the LAN port instead
- T38Enable Off
- Using PJSIP and port instead (same issue)
- Force g729 codec instead
- Increase SilenceTimeThreshold
- DetectFarEndLongSilence Off
- DetectCPC Off
- SilenceDetectSensitivity to Low
- Disabled registration
- Adjusted MTU size

As mentioned watching the call status, packet counts continue to increase during the one-way audio dropout. Codecs remain as specified (711u) and there isn't anything abnormal in the displayed info (such as increased dropped packets, late packets, etc.).

It seems like the dropped audio problem happens more quickly when calling / receiving calls from cellular phones than from pots lines but this may be a coincidence.


drgeoff

1.  A router actions SIP ALG only on WAN traffic.  Your OBi to PBX packets are purely on the LAN.

2.  Are you certain about where the incoming audio is getting dropped?  If you are using an analogue phone on the 212 then you have

POTS LINE -- OBi212 LINE jack -- LAN -- PBX -- LAN -- OBi212 phone jack -- phone.

If an incoming POTS call goes to voicemail, are there drops in the recorded message?

Are the drops in incoming audio present in both POTS calls made and POTS calls received or in only one of those?

If you use the analogue phone on calls to and from a VoIP distant end do the drops occur?

Nmennear


Nmennear

Quote from: drgeoff on March 29, 2019, 12:39:16 PM
1.  A router actions SIP ALG only on WAN traffic.  Your OBi to PBX packets are purely on the LAN.

2.  Are you certain about where the incoming audio is getting dropped?  If you are using an analogue phone on the 212 then you have

POTS LINE -- OBi212 LINE jack -- LAN -- PBX -- LAN -- OBi212 phone jack -- phone.

If an incoming POTS call goes to voicemail, are there drops in the recorded message?

Are the drops in incoming audio present in both POTS calls made and POTS calls received or in only one of those?

If you use the analogue phone on calls to and from a VoIP distant end do the drops occur?

1. Thanks, this makes sense

2. The dropped audio is always on my side but we're going to test based on your scenarios and i will report back. Our configuration is:

POTS LINE -- OBi212 LINE jack -- LAN -- PBX -- LAN -- Yealink t46s SIP endpoint

Calls between extensions (entirely internal calls) are working perfectly.

I will report back soon, thanks for your suggestions!

UPDATE: I've done the above tests and calls going to VM, pots calls made and pots calls received all have the audio drops. Any call made through FreePBX over the pstn has the same intermittent issue on the incoming audio (only).

As an aside, testing directly at the pots demarc shows it's not an issue at that point. Likewise, i made a couple calls from an analog phone attached to the OBi FXS port (directly to the pots line / bypassing the FreePBX entirely) which seem to not have the intermittent audio issue either.

My testing has shown the intermittent audio can start almost immediately after the start of the call or can take 5-10+ minutes to start.

buffaloco

Your issue sounds almost exactly like the issue that we are experiencing and have been troubleshooting. Did you ever find any resolution to the issue?

drgeoff

Quote from: Nmennear on March 29, 2019, 01:02:18 PM
Quote from: drgeoff on March 29, 2019, 12:39:16 PM
1.  A router actions SIP ALG only on WAN traffic.  Your OBi to PBX packets are purely on the LAN.

2.  Are you certain about where the incoming audio is getting dropped?  If you are using an analogue phone on the 212 then you have

POTS LINE -- OBi212 LINE jack -- LAN -- PBX -- LAN -- OBi212 phone jack -- phone.

If an incoming POTS call goes to voicemail, are there drops in the recorded message?

Are the drops in incoming audio present in both POTS calls made and POTS calls received or in only one of those?

If you use the analogue phone on calls to and from a VoIP distant end do the drops occur?

1. Thanks, this makes sense

2. The dropped audio is always on my side but we're going to test based on your scenarios and i will report back. Our configuration is:

POTS LINE -- OBi212 LINE jack -- LAN -- PBX -- LAN -- Yealink t46s SIP endpoint

Calls between extensions (entirely internal calls) are working perfectly.

I will report back soon, thanks for your suggestions!

UPDATE: I've done the above tests and calls going to VM, pots calls made and pots calls received all have the audio drops. Any call made through FreePBX over the pstn has the same intermittent issue on the incoming audio (only).

As an aside, testing directly at the pots demarc shows it's not an issue at that point. Likewise, i made a couple calls from an analog phone attached to the OBi FXS port (directly to the pots line / bypassing the FreePBX entirely) which seem to not have the intermittent audio issue either.

My testing has shown the intermittent audio can start almost immediately after the start of the call or can take 5-10+ minutes to start.
I hadn't noticed your update until just now.

That you can make and receive calls OK using analogue phone - OBi212 - POTS suggests that all the analogue gubbins and some of the digital gubbins of the 212 is not responsible for the audio drops.  But you do not seem to have tested digital/IP stuff from 212 to the PBX.  Make a call from the analogue phone on the 212 to an extension on the PBX.  And make a call from the analogue phone on the 212 using VoIP but not going through the PBX.  That is using an SP on the 212 that is not the PBX and not POTS.  If you don't have an ITSP that you can temporarily register one of the 212's SPs to, try the OBi test number **9 222 222 222.

If the first of those tests has the drops, that would point to the 212 to PBX section.  If the second one never drops incoming audio that would point to the issue being more with the PBX, whereas if it does have drops it would point to a problem with the 212.